Thesis (M.S.)--Brigham Young University. Dept. of Mechanical Engineering, 2005. / Includes bibliographical references.

In the ocean, changes in the speed of sound can be related to changes in water temperature. By leveraging this relationship, acoustic methods – namely acoustic tomography- have been used to monitor temperature changes in the deep ocean for the purposes of providing inputs to climate change models. Traditionally, these acoustic methods involve loud, active sound sources which can be logistically challenging to operate and have been criticized for potentially disturbing marine animals. Therefore, this work demonstrates a passive acoustic method - previously only used in shallow water for short monitoring durations- that uses only recordings of low-frequency (1-40 Hz) ambient noise to continuously monitor variations in deep ocean temperature with an unprecedented degree of precision and temporal resolution. Numerical simulations were conducted to show the portions of the ocean that are monitored with this passive method. This work also provides recommendations (regarding sensor placement around the world) for future development of a global passive acoustic sensor network that makes use of distant noise sources (sea-ice or seismic sources) to extract meaningful information (whether temperature, currents, etc.) about the ocean. Finally, an optimization method is proposed to overcome one of the fundamental limitations of previous applications of this passive monitoring method: tracking oceanic fluctuations that occur over short time scales. Hence, the results of this study may assist in the development of more reliable climate models that include an enhanced understanding of the ocean’s role as a global heat sink. Finally, an optimization method was proposed to enhance the emergence rate of coherent arrivals from ambient noise correlations, thus allowing this passive monitoring method to track acoustic medium fluctuations on a shorter time scale. This optimization could also be used in other applications of noise-based passive monitoring in a rapidly fluctuating medium (seismic, structural health monitoring, biomedical, etc.).

The spatial evolution of acoustic waveforms produced by a Mach 3 jet are investigated using both 1/4 inch and 1/8 inch pressure field microphones located along rays emanating from the post potential core where the peak sound emission is found to occur. The measurements are acquired in a fully anechoic chamber where ground, or other large surface reflections are minimal. The calculation of the OASPL along an arc located at 95 jet diameters using 120 planar grid measurements are shown to collapse remarkably well when the arc array is centered on the post potential core region. Various statistical metrics, including the quadrature spectral density, number of zero crossings, the skewness of the pressure time derivative and the integral of the negative part of the quadrature spectral density, are exercised along the peak emission path. These metrics are shown to undergo rapid changes within 2 meters from the source regions of this laboratory scale jet. The sensitivity of these findings to both transducer size and humidity effects are discussed. A visual extrapolation of these nonlinear metrics toward the jet shear layer suggests that these waveforms are initially skewed at the source. An experimentally validated wave packet model is used to confirm the location where the pressure decay law transition from cylindrical to spherical. It is then used to estimate the source intensity, which is required to predict the effective Gol'dberg number. / text

No description available.

The backpropagation method of image reconstruction has been known for some time with the advantage of fast processing due to the use of Fast Fourier Transform. But its applicability to underwater imaging has been limited. At present the shift-and-add method is the more widely used method in underwater imaging. This is due to the fact that backpropagation has been derived for plane wave insonification, with the scattered waves detected in transmission-mode, or synthetic aperture set-up. One of the methods being used for underwater imaging is to use a point source for the insonification of the target and the scattered waves detected in reflection-mode by a receiver array. An advantage of this scanning method is only one transmission of the source is required to capture an image, instead of multiple transmissions. Therefore motion artifacts are kept to minimum. To be able to exploit the processing speed of the backpropagation method, it must be adapted for point source insonification. The coverage of this configuration in the literature has been scant, methods for spherical sources have been proposed for transmission mode and arbitrary surfaces in geophysical applications. These methods are complex and difficult to use. A novel point source compensation method is proposed in this thesis so that the backpropagation image formation method can be used for the point source insonification set-up. The method of investigation undertaken to derive this new backpropagation method was through theoretical analysis, numerical simulation and experimental verification. The effect of various compensation factors on the image quality was studied in simulation. In the experimental verification, practical issues relating to the application of the new method was addressed. The final proof of concept of our method was undertaken with our experimental verification. The quality of images formed with the point source compensation methods has also been compared with that with the shiftand- add method. Experimental and simulation results show that the point source compensated backpropagation algorithm can produce images of comparable quality with those formed with shift-and-add method for the set-up of wideband point-source insonification with detection in reflection-mode, with the advantage of faster image formation.

No description available.

Thesis (Ph. D.)--University of Texas at Austin, 2003. / Vita. Includes bibliographical references.

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The ability to accurately manipulate the spectral content of audio is indispensable for the artistic presentation of sound. While there are many devices currently available that are capable of performing this function, most tend to be either highly complex electrical engineering tools, or music-oriented products that are limited in functionality. Filter Designer was created to fill this gap by providing an environment with which musically-trained users can design and implement digital filters, while having access to control parameters and analysis data previously exclusive to the engineering field. This work explores the factors that motivated the creation of Filter Designer, and examines the process of its development, from basic user interface design to the calculation and implementation of digital filters for use with audio signals.

Filters constitute an essential tool for manipulating the spectral content of a signal. While there is a plethora of filtering tools, both in the hardware and software domain, the majority of them are geared towards engineers and scientists, rather than sound designers and electroacoustic composers. The "common-practice" approach is to consider filters as post-production tools. This can be restrictive if filters are to be used as artistic tools, dynamically involved in the shaping of the sound. This thesis was written with this approach in mind its aim is (a) to provide a survey of the various digital recursive filters, enabling a filter designer to choose the one that suits his needs, (b) to teach filter designers, such as electroacoustic composers and sound designers how to calculate digital filter coefficients, and (c) implement filter algorithms using the familiar syntax of computer music languages such as Csound and SuperCollider .